[Asterisk-Users] vmail.cgi question
When I log into the web page to get my voicemail I see that there nothing being listed. I change the following line in vmail.cgi so I do not need to login with my extension plus context. $context="local"; # Define here your by default context (so you dont need to put [EMAIL PROTECTED] in the login I also created a new symbolic link to point to local direct instead to default: lrwxrwxrwx 1 root root 35 Jul 18 11:01 vm -> /var/spool/asterisk/voicemail/local Am I missing something else. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Comedian Web page login
When I try to login into voicemail through the web interface It states incorrect login. In my voicemail.conf I have all voicemail boxes set under local. I changed the symbolic link to reflect the new directory under /var/spool/asterisk. Am I missing something? My vm link = /var/spool/asterisk/voicemail/local. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Reliable Provider
Net2phone Kurt __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ATA registration requests
I set the SISIPRegIntervalo 3600. Wouldn't that mean to send a registration packet every hour instead of every miminuter so. --OR-- Is this the typical reresponseack to the * server when the ATATAeceive a SIP Notify. Kurt __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA registration requests
I have two ATA186 running 2.14 and 2.15. I see in the SIP debugs that both ATAs keep on sending SIP registration packets over and over. The flow is as follows: Asterisk receives REGISTER packet Asterisk sends 100 trying, 200 ok with an expire of 3600. Kurt __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sacking calls to extension to voicemail
A SIP call comes into the * server on a number that I want to immediately sack to vovoicemail. How would this be achieved Kurt __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Voice Mail Integration with Cisco CME
Since the voice mail portion of CME is an additional charge, I was wondering has any body use Asterisk voice mail with Cisco CME. Kurt. __ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
